Commit Graph

69712 Commits

Author SHA1 Message Date
Andres Salomon
4ea2416398 [ALSA] cs5535audio: drop unused bus master stuff
We really only care about the first two bus masters (playback and capture).
There's no need to have unused BM code lying around, so let's get rid of it.
If for some reason we trigger an IRQ for some BM that we're not using.. well,
that warrants spitting out an error message (imo).

Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:54 +02:00
Andres Salomon
506ea68cd9 [ALSA] cs5535audio: fix ACC_BM[x]_CMD register handling
According to 6.3.2.7 of the cs5535/cs5536 data sheets, the ACC_BM[x]_CMD
registers are only 8 bits wide.  This driver treats them as 32 bits wide,
and also has bits in the wrong place.  Simple fix to the definitions.

Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:53 +02:00
Andres Salomon
1caae3682e [ALSA] cs5535audio: update PCI device handling in suspend/resume
Save the PCI state before disabling the device, and add some error checking.

Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:53 +02:00
Andres Salomon
222fa0b0d2 [ALSA] cs5535audio: fix PRD register save/restore power management race
In the suspend path, we currently save the PRD registers and then disable DMA.
This is racy; the sound hardware might update the PRD register as it finishes
processing some DMA pages between when we've saved the PRD registers and
when DMA actually gets disabled.  Furthermore, we actively check whether or
not DMA is enabled before saving PRD registers; there's no reason to do that,
as the PRD registers should not update when we twiddle the ACC_BM[x]_CMD
register(s).  Worst case, we save the PRD registers twice; even powering
down the ACC shouldn't mess with the PRD registers (according to the 5536
data sheet, section 5.3.7.4, power-down procedure).  This patch reworks
all that to first disable DMA, and then save PRD registers.

Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:52 +02:00
Andres Salomon
7abcacb09a [ALSA] cs5535audio: correctly set dma->substream
We're never actually setting dma->substream to the current substream; that
means the dma->substream checks that we do in the suspend/resume path
are never satisfied, and the PRD registers are never correctly managed.  This
changes it so that we set the substream when constructing the specific
bus master DMA, and unsetting it when we tear down the BM's DMA.

Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:51 +02:00
Maxim Levitsky
9e05b7a3d9 [ALSA] hda-codec - Fix support for sigmatel codecs that have 2 or more ADCs
1) Create seperate mixer controls for each ADC
2) Make number of substreams of capture PCM device be equal to
   number of ADCs

Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:50 +02:00
Maxim Levitsky
6e6b88ffea [ALSA] hda-codec - make volume knob, the master volume for sigmatel codecs
VolumeKnob is present on most sigmatel codecs, it allows to decrease
volume of all DACs at once, it is a kind of post-procesing volume.
Note that all output amps of sigmatel only decrease volume, and all
input amps only increase volume.

Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:50 +02:00
Maxim Levitsky
5f10c4a9a0 [ALSA] hda-codec - add support for analog loopback to STAC9204/9205/922x/927x
The analog loopback routes the sound just before it enters ADC0
to output of DAC0.

Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:49 +02:00
Maxim Levitsky
0fb87bb474 [ALSA] hda-codec - add support for swapping center/LFE channels to STAC codecs
Center/LFE channels are located on same jack, so it can be usefull
to swap them.

Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:48 +02:00
Maxim Levitsky
d804ad9258 [ALSA] hda-intel - Fix resume logic, when dynamic power managment is on
Comment in hda_intel.c states that 'the explicit resume is needed only
when POWER_SAVE isn't set', but this is not true.
There is no code that will automaticly power up the codec on resume,
but only code that powers it up when user accesses it. So if user
leaves a sound playing, codec will not be powered
To fix that I check if there are any codecs that should be powered
codec->power_count, and if so I power them up together with main
controller.

Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:47 +02:00
Maxim Levitsky
2e4924628a [ALSA] hda-intel - fix a race in dynamic power managment
codec->power_transition is supposed to be true while codec is going
to be shut off if in the mean time somebody calls snd_hda_power_up,
hda_power_work will not shut down the codec, but nether will clear
codec->power_transition, thus it stays on forever. Fix this.

Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:46 +02:00
Clemens Ladisch
b7e054a76f [ALSA] cmipci: show real chip name in card name
The '-MCx' suffix that is expected by alsa-lib is only needed in the
card driver string, so we can show the actual chip name in the
shortname.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:46 +02:00
Clemens Ladisch
88039815d8 [ALSA] cmipci: check that the legacy MIDI port works
Check that the UART_EN bit actually enabled the MPU-401 port.
Apparently, C-Media thinks that it is a good idea to be paranoid here.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:45 +02:00
Clemens Ladisch
c78c950d28 [ALSA] cmipci: do not check for integrated FM/MIDI ports with chip version 37
Integrated MPU-401/OPL3 ports are available with chip version 39 and
later, so we do not test for the port with version 37.
Now that the test is known to work, we can again enable the MIDI port by
default.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:44 +02:00
Clemens Ladisch
8992e18db3 [ALSA] cmipci: add 96 kHz support
Add support for 88.2 kHz and 96 kHz analog and digital playback on
CMI8768/CMI8770 chips.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:43 +02:00
Clemens Ladisch
f19a82a119 [ALSA] cmipci: remove invalid channels constraint
Remove the constraint that sets the channel limit for the first playback
device to that of the second one; the first device supports only stereo.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:42 +02:00
Takashi Iwai
c480f79bdc [ALSA] hda-codec - Avoid zero NID in line_out_pins[] of STAC codecs
The STAC codes adds line_out_pins[] for shared mic/line-inputs accordingly.
But, the current code may give a hole with NID=0 in some setting, which
results in an error at probe.  This patch fixes the problem.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:41 +02:00
Takashi Iwai
dc81bed127 [ALSA] hda-codec - Fix wrong pin-setup at resume of STAC codecs
The resume procedure for STAC codecs overrides the cached values and
results in the wrong (reset) PIN state.  The patch gets rid of the
overriding part and simplifies the resume.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:40 +02:00
Mark Hills
3a7788b751 [ALSA] usb-audio: update quirk for Rane SL 1 (aka. Serato Scratch Live)
Allow the interface's mixer to be used, and give the interface its
correct name.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:40 +02:00
Maxim Levitsky
ca7c5a8b4b [ALSA] hda-codec - code cleanups in patch_sigmatel.c
Clean up the mixer entries for Input Source using a macro.

Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:39 +02:00
zhejiang
accbe4988c [ALSA] hda-codec - Fix capture on ALC262 HP machines
Fix the index for Front Mic capture source on ALC262 HP machines.
Also, added the new capture source list for HP BPC DC7000 series
to work properly.
From: zhejiang <zhe.jiang@intel.com>

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:38 +02:00
Remy Bruno
a3a68c85bf [ALSA] hdsp - Add support for latset RME9632 revisions
added support for the latest revision of the 9632 (and hopefully a few
following ones). The DSP matrix was not working because of wrong
identification of the card in this part of the code.

Signed-off-by: Remy Bruno <remy.bruno@trinnov.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:37 +02:00
Remy Bruno
6534599d14 [ALSA] hdspm - Fix autosync bug
* better report of speed mode change failures
* autosync_ref control bugfix (was reporting pref_sync_ref instead)
  (changed HDSPM_AES32_AUTOSYNC_FROM_NONE value to comply with array
  indexing in snd_hdspm_info_autosync_ref())
* added support for master modes up to 192kHz (clock source control
  value was restricted up to 96kHz)

Signed-off-by: Remy Bruno <remy.bruno@trinnov.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:37 +02:00
Oliver Neukum
5149fe2c15 [ALSA] missing error check in usb sound driver
usb_set_interface() can fail, even for altsetting 0

Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:36 +02:00
Clemens Ladisch
15944806e2 [ALSA] usb-audio: add quirk for Serato Scratch Live DJ Box
Add a quirk to detect the Serato Scratch Live DJ Box.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-10-16 15:59:35 +02:00